IP networks generally provide an excellent infrastructure for geographically distributing components of a telecommunication system. The underlying IP network is optimal for transmission for control signaling, and when bandwidth is available, can provide an acceptable quality of service (or QoS) or grade of service (or GoS) for voice communications. When insufficient network resources are available for voice communications or one or more IP network components are down, voice communications can be adversely impacted.
A number of techniques have been attempted to address these issues.
In one technique, if a system had multiple communication gateways controlled by a single controller and the private switching facilities inter-connecting these gateways failed, users can “dial-out” on a public network trunk using the public address (or Direct Inward Dialing or DID number) of the destination party. This approach requires manual intervention by the user first to recognize that a problem exists, second to determine how to circumvent it, and third to dial the DID number. Normally, the calling party would dial only an extension to reach the destination party. If the destination party to be reached does not have a public number, he or she is not reachable by the alternate network.
In another technique known as PSTN Fallback™ of Avaya Inc., a call is forced to the PSTN when an IP trunk connection experiences an unacceptable QoS or GOS. In particular, in a multi-enterprise architecture, each enterprise may have a separate, independent, and active or primary media servers with resident call controller functionality, a plurality of digital stations, a plurality of IP or Internet Protocol stations, a gateway and a Local Area Network or LAN. The media servers are independent in that one media server in one enterprise is generally unaware of the subscriber configuration information, such as extensions, of the other enterprise's subscribers. The gateways are interconnected by the Public Switched Telephone Network or PSTN and Wide Area Network or WAN. When a call is to be placed over the WAN, the originating call controller determines the currently measured network delay and packet loss. When either measured variable reaches a predetermined threshold, the call controller automatically takes the idle IP trunk ports out-of-service, i.e., it busies out the ports. The ports remain out-of-service until the measurements return to the low threshold. No new calls are allowed over the IP trunk. Normal or conventional call routing over the PSTN is used for access to the next preference in the route pattern.
In another technique known as Separation of Bearer and Signaling™ (SBS) of Avaya Inc., the signaling channel for a call is routed over the WAN while the bearer channel is routed over the PSTN. The signaling channel in SBS includes SBS call-control signaling and QSIG private-networking protocol information. SBS associates the signaling and bearer channels at the SBS originating and terminating nodes so that they appear to the end users to be a normal, non-separated call. The use of the WAN for signaling traffic and the PSTN for voice bearer traffic addresses a customer need for using small amounts of bandwidth in the IP WAN for signaling traffic, with the voice bearer portion of the call being sent over inexpensive PSTN facilities.
PSTN Fallback™ and SBS™ address architectures where there exist multiple, separate system implementations inter-connected by a traditional inter-switch trunking protocol; in other words, they permit inter-connection only of peer-to-peer systems. With the move to larger, single-server IP WAN-connected media gateway distributed systems, there is no longer a need for IP trunks and SBS. Using trunk group administration to limit bandwidth between media servers is not required nor is PSTN Fallback™ when the number of calls exceeds the administered IP trunk member limit. There is no need to embed an intelligent signaling interface between servers over IP WAN facilities given that the system has only a single active or primary server and that all calls across the system appear to be station-to-station calls.
Another technique for managing IP bandwidth usage includes call admission control in which the number of calls across the WAN or the bandwidth available for voice calls is limited. Call admission control can result in the call being denied and being forwarded to the callee's voice mail server (if accessible), thereby causing caller frustration.
In other systems, a communication manager providing a call admission control function can perform alternate routing, but not over the PSTN. In still other systems, alternate routing can be performed over the PSTN, but not on behalf of session initiation protocol (SIP) endpoints because such endpoints are not controlled by such communication servers. In these systems, the communication server merely functions as a feature server for the SIP endpoints.
These systems typically effect alternate routing over the PSTN using dual tone multiple frequency (DTMF) signaling techniques. However, such techniques are slow.
Accordingly, there is a need for systems and methods that provide for routing a call between SIP endpoints when the primary wide area network (WAN) is too busy to provide satisfactory quality of service and/or bandwidth.